The present invention is directed to real time oscilloscopes, and more particularly to real time oscilloscopes employing a bandwidth extension scheme (i.e. Digital Bandwidth Interleaving), such as that described in U.S. patent application Ser. No. 11/281,075, titled HIGH BANDWIDTH OSCILLOSCOPE, filed Nov. 17, 2005 (“the '075 application”), currently pending, the entire contents thereof being incorporated herein by reference. In accordance with the disclosure of this pending patent application, each of a plurality of channels (two, for example, which will be referred to as LF (low frequency) and HF(high frequency)) receive a portion of an input signal corresponding to a specific frequency band. As described in the above application, additional processing is performed on this split signal for acquiring and digitizing the signal. In accordance with the present invention, further modification to this process of the '075 application is presented.
The DBI technique referenced above allows extending the bandwidth of Digital Storage Oscilloscopes (DSO's) by using bandwidth available in additional channels of the DSO to augment the bandwidth of a primary channel or channels. The application of such a DBI technique is limited not only by the bandwidth available on each channel but also by the necessity to leave sufficient distance between the bands for upsampling. Also, to allow for the use of the greatest amount of theoretically available bandwidth, the cross-over regions between adjacent bands have to be narrow, causing various challenges such as large group delay variations.
Ideally, two bands of width fS/2, one extending between 0 and fS/2 and the other extending between fS/2 and fS, can be combined into a channel of width fS and sampling rate 2 fS. However, such a scheme appears to be physically impossible because no hardware filter structure can be made to transmit 100% frequencies below fS/2, and transmit 0% frequencies above fS/2, and vice versa. Furthermore, software filters encounter a similar challenge: no finite-length linear digital filter can upsample(interpolate) leaving the original frequency 100% intact and create 0% spur, when the frequency is nearly equal to Nyquist.
The invention is a method for setting the imperfections of software interpolation filter to be commensurate with the hardware filter imperfections. When the samplings of the channel connected with the low frequency are staggered in between the samplings of the high frequency channel, and the interpolation filter taps are set equal to the other channel's input response scaled values, a substantially perfect cancellation (to the level of precision of the instrument) of terms appearing at the wrong frequency can be achieved in accordance with the invention. This is analogous to the well-known aliasing-canceling condition of filterbanks, and the description will show similarities and differences between the present invention and this specific digital filtering technique.
As presented in the '075 application, fLO represents the frequency of the mixer's LO (element 43 of FIG. 2 in the '075 application). fS represents the sampling frequency of the digitizers (elements 5, 6, 7, 8 of FIG. 1 in the '075 application). In such a system, fLO would be equal to fS, with half the bandwidth [0 to fS/2] going to the LF channel and half the bandwidth [fS/2 to fS] going to the HF channel. As noted above, it has been determined by the inventor of the present invention that when one tries to build such a system employing the maximum theoretical bandwidth, two phenomena impede optimal operation, namely the width of the hardware filters and the width of the software interpolation filters. In accordance with the present invention, the inventor herein presents a technique that circumvents these limitations.